Commit Graph

811 Commits

Author SHA1 Message Date
Eldad Zack
fee4b700a4 ALSA: hiface: return correct XRUN indication
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of hiface, as expected by snd_pcm_update_hw_ptr0().

Caught by sparse.

Cc: Antonio Ospite <ospite@studenti.unina.it>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-24 10:51:37 +02:00
Eldad Zack
be2f93a4c4 ALSA: usb-audio: 6fire: return correct XRUN indication
Return SNDRV_PCM_POS_XRUN (snd_pcm_uframes_t) instead of
SNDRV_PCM_STATE_XRUN (snd_pcm_state_t) from the pointer
function of 6fire, as expected by snd_pcm_update_hw_ptr0().

Caught by sparse.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-21 11:55:22 +02:00
Takashi Iwai
5be1efb4c2 ALSA: usx2y: Fix unlocked snd_pcm_stop() call
snd_pcm_stop() must be called in the PCM substream lock context.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15 21:25:13 +02:00
Takashi Iwai
9538aa46c2 ALSA: ua101: Fix unlocked snd_pcm_stop() call
snd_pcm_stop() must be called in the PCM substream lock context.

Cc: <stable@vger.kernel.org>
Acked-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15 21:24:57 +02:00
Takashi Iwai
5b9ab3f732 ALSA: 6fire: Fix unlocked snd_pcm_stop() call
snd_pcm_stop() must be called in the PCM substream lock context.

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-15 18:12:50 +02:00
Eldad Zack
42d4ab832d ALSA: usb-audio: fix regression for fixed stream quirk
Commit 8f898e92ae removed the redundant
reads of bInterfaceProtocol from the descriptors, but introduced a
regression to devices with quirks of type QUIRK_AUDIO_FIXED_ENDPOINT,
since fp->protocol is not set in setup process.

As a consequence, audio streams would not get initialized, as the
following logs show:

[   48.923043] setting usb interface 3:1
[   48.923056] Creating new capture data endpoint #81
[   48.923484] 4:3:1: cannot set freq 48000 to ep 0x81

This patch sets fp->protocol in create_fixed_stream_quirk() and
resolves the regression.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-07-10 17:52:14 +02:00
Przemek Rudy
066624c6a1 ALSA: usb-audio: Add Audio Advantage Micro II
This patch is adding extensive support (beside standard usb audio class)
for Audio Advantage Micro II usb sound card.
Features included:
- Access to AES bits (so now sending the IEC61937 compliant stream is
possible).
- Mixer SPDIF control added to turn on/off the optical transmitter.

Signed-off-by: Przemek Rudy <prudy1@o2.pl>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-28 13:37:12 +02:00
Takashi Iwai
ea70ee057c Merge branch 'full-roland-support' of git://git.alsa-project.org/alsa-kprivate into for-next
For adding support for many Roland and Yamaha devices:
* 'full-roland-support' of git://git.alsa-project.org/alsa-kprivate:
  ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
  ALSA: usb-audio: claim autodetected PCM interfaces all at once
  ALSA: usb-audio: remove superfluous Roland quirks
  ALSA: usb-audio: add MIDI port names for some Roland devices
  ALSA: usb-audio: add support for many Roland/Yamaha devices
  ALSA: usb-audio: detect implicit feedback on Roland devices
  ALSA: usb-audio: store protocol version in struct audioformat
2013-06-28 12:13:26 +02:00
Clemens Ladisch
b7f33917bc ALSA: usb-audio: add quirks for Roland QUAD/OCTO-CAPTURE
The Roland Quad/Octo-Capture devices use some unknown vendor-specific
mechanism to switch sample rates (and to manage other controls).  To
prevent the driver from attempting to use any other than the default
44.1 kHz sample rate, use quirks to hide the other alternate settings.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:50 +02:00
Clemens Ladisch
b1ce7ba619 ALSA: usb-audio: claim autodetected PCM interfaces all at once
snd_card_register() registers all devices newly added since the last
call.  However, the playback/capture streams are handled as one ALSA
device, so the second /dev device will not be registered if the PCM
streams are added in two steps.

QUIRK_AUTODETECT caused the probe callback to be called once for each
interface, which triggered this problem.  Work around this by handling
this like the composite quirk, i.e., autodetecting all other interfaces
that might be used for PCM or MIDI.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:49 +02:00
Clemens Ladisch
8e5ced83dd ALSA: usb-audio: remove superfluous Roland quirks
Remove all quirks that are no longer needed now that the generic Roland
quirks can handle the vendor-specific descriptors correctly.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:49 +02:00
Clemens Ladisch
a968782e27 ALSA: usb-audio: add MIDI port names for some Roland devices
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:48 +02:00
Clemens Ladisch
aafe77cc45 ALSA: usb-audio: add support for many Roland/Yamaha devices
Add quirks to detect the various vendor-specific descriptors used by
Roland and Yamaha in most of their recent USB audio and MIDI devices.

Together with the previous patch, this should add audio/MIDI support for
the following USB devices:
- Edirol motion dive .tokyo performance package
- Roland MC-808 Synthesizer
- Roland BK-7m Synthesizer
- Roland VIMA JM-5/8 Synthesizer
- Roland SP-555 Sequencer
- Roland V-Synth GT Synthesizer
- Roland Music Atelier AT-75/100/300/350C/500/800/900/900C Organ
- Edirol V-Mixer M-200i/300/380/400/480/R-1000
- BOSS GT-10B Effects Processor
- Roland Fantom G6/G7/G8 Keyboard
- Cakewalk Sonar V-Studio 20/100/700 Audio Interface
- Roland GW-8 Keyboard
- Roland AX-Synth Keyboard
- Roland JUNO-Di/STAGE/Gi Keyboard
- Roland VB-99 Effects Processor
- Cakewalk UM-2G MIDI Interface
- Roland A-500S Keyboard
- Roland SD-50 Synthesizer
- Roland OCTAPAD SPD-30 Controller
- Roland Lucina AX-09 Synthesizer
- BOSS BR-800 Digital Recorder
- Roland DUO/TRI-CAPTURE (EX) Audio Interface
- BOSS RC-300 Loop Station
- Roland JUPITER-50/80 Keyboard
- Roland R-26 Recorder
- Roland SPD-SX Controller
- BOSS JS-10 Audio Player
- Roland TD-11/15/30 Drum Module
- Roland A-49/88 Keyboard
- Roland INTEGRA-7 Synthesizer
- Roland R-88 Recorder

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:48 +02:00
Clemens Ladisch
ba7c2be114 ALSA: usb-audio: detect implicit feedback on Roland devices
All the Roland/Edirol/BOSS USB audio devices that need implicit feedback
show this unambiguously in their descriptors, so it might be a good idea
to let the driver detect this.

This should make playback work correctly (at least with Jack) with the
following devices:
- BOSS GT-100
- BOSS JS-8 Jam Station
- Edirol M-16DX
- Roland GAIA SH-01

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:47 +02:00
Clemens Ladisch
8f898e92ae ALSA: usb-audio: store protocol version in struct audioformat
Instead of reading bInterfaceProtocol from the descriptor whenever it's
needed, store this value in the audioformat structure.  Besides
simplifying some code, this will allow us to correctly handle vendor-
specific devices where the descriptors are marked with other values.

Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
2013-06-27 21:59:47 +02:00
Antonio Ospite
a91c3fb2f8 Add M2Tech hiFace USB-SPDIF driver
Add driver for M2Tech hiFace USB-SPDIF interface and compatible devices.

M2Tech hiFace and compatible devices offer a Hi-End S/PDIF Output
Interface, see http://www.m2tech.biz/hiface.html

The supported products are:

  * M2Tech Young
  * M2Tech hiFace
  * M2Tech North Star
  * M2Tech W4S Young
  * M2Tech Corrson
  * M2Tech AUDIA
  * M2Tech SL Audio
  * M2Tech Empirical
  * M2Tech Rockna
  * M2Tech Pathos
  * M2Tech Metronome
  * M2Tech CAD
  * M2Tech Audio Esclusive
  * M2Tech Rotel
  * M2Tech Eeaudio
  * The Chord Company CHORD
  * AVA Group A/S Vitus

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-24 09:26:08 +02:00
Antonio Ospite
0af49ffe3c ALSA: usb: uniform style used in MODULE_SUPPORTED_DEVICE()
In sound/usb/card.c and sound/usb/misc/ua101.c there are no spaces
between the vendor and the device names, use this style in the other
drivers too.

This also helps keeping consistency when new drivers copies from the
ones already in the mainline tree.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:37:08 +02:00
Antonio Ospite
4a9f911861 ALSA: snd-usb-6fire: use vmalloc buffers
For USB devices it's not necessary to allocate physically contiguous
buffers.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:36:41 +02:00
Antonio Ospite
fc76f86376 ALSA: snd-usb-caiaq: use vmalloc buffers
For USB devices it's not necessary to allocate physically contiguous
buffers.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:35:52 +02:00
Antonio Ospite
3dd446a7e5 ALSA: snd-usb-caiaq: remove the unused snd_card_used variable
The snd_card_used variable is only read but never written, remove it.

Signed-off-by: Antonio Ospite <ao2@amarulasolutions.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-21 14:33:05 +02:00
Dave Jones
cd1199edc7 ALSA: sound/usb/misc/ua101.c: convert __list_for_each usage to list_for_each
Signed-off-by: Dave Jones <davej@redhat.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-18 07:47:32 +02:00
Dan Carpenter
da177dd025 ALSA: usx2y: remove some old dead code
USB_QUEUE_BULK isn't defined any more.

Signed-off-by: Dan Carpenter <dan.carpenter@oracle.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17 10:45:42 +02:00
Takashi Iwai
36691e1be6 ALSA: usb-audio: Fix invalid volume resolution for Logitech HD Webcam c310
Just like the previous fix for LogitechHD Webcam c270 in commit
11e7064f35, c310 model also requires the
same workaround for avoiding the kernel warning.

Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=59741
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17 10:25:02 +02:00
Clemens Ladisch
342cda2934 ALSA: usb-audio: work around Android accessory firmware bug
When the Android firmware enables the audio interfaces in accessory
mode, it always declares in the control interface's baInterfaceNr array
that interfaces 0 and 1 belong to the audio function.  However, the
accessory interface itself, if also enabled, already is at index 0 and
shifts the actual audio interface numbers to 1 and 2, which prevents the
PCM streaming interface from being seen by the host driver.

To get the PCM interface interface to work, detect when the descriptors
point to the (for this driver useless) accessory interface, and redirect
to the correct one.

Reported-by: Jeremy Rosen <jeremy.rosen@openwide.fr>
Tested-by: Jeremy Rosen <jeremy.rosen@openwide.fr>
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-17 09:56:52 +02:00
Takashi Iwai
11e7064f35 ALSA: usb-audio - Fix invalid volume resolution on Logitech HD webcam c270
USB audio driver spews an error message when probing Logitech HD
webcam c270:
  ALSA mixer.c:1300 usb_audio: Warning! Unlikely big volume range (=6144), cval->res is probably wrong.
  ALSA mixer.c:1304 usb_audio: [5] FU [Mic Capture Volume] ch = 1, val = 1536/7680/1

Obviously the device needs a fixed volume resolution (cval->res = 384)
like other Logitech devices.

Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=821735

Reported-and-tested-by: Cristian Rodríguez <crrodriguez@opensuse.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-05 08:35:26 +02:00
Takashi Iwai
8eafc0a161 ALSA: usb-audio - Apply Logitech QuickCam Pro 9000 quirk only to audio iface
... instead of applying to all interfaces.

Reference: http://forums.gentoo.org/viewtopic-p-6886404.html

Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-04 16:07:48 +02:00
Clemens Ladisch
a0c6d309c6 ALSA: usb-audio: fix Roland/Cakewalk UM-3G support
Commit 927c9423dd (ALSA: usb-audio: add
Edirol UM-3G support) used a wrong quirk type, which would make the
driver refuse to attach with the error message "MIDIStreaming interface
descriptor not found".

Cc: <stable@vger.kernel.org> # 3.3 and later
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-06-03 09:42:21 +02:00
Torsten Schenk
d47333ddb2 ALSA: usb-6fire: Modify firmware version check
Check only the uppermost 16 bits instead of the whole 32 bits of
the version information. Do this because all firmware version tested
with this version information worked correctly and the strict check
causes problems for several users.

Signed-off-by: Torsten Schenk <torsten.schenk@zoho.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-23 14:30:26 +02:00
Torstein Hegge
e6135fe960 ALSA: usb-audio: proc: use found syncmaxsize to determine feedback format
freqshift is only set for the data endpoint and syncmaxsize is only set
for the sync endpoint. This results in a syncmaxsize of zero used in the
proc output feedback format calculation, which gives a feedback format
incorrectly shown as 8.16 for UAC2 devices.

As neither the data nor the sync endpoint gives all the relevant
content, output the two combined.

Also remove the sync_endpoint "packet size" which is always zero
and the sync_endpoint "momentary freq" which is constant.

Tested with UAC2 async and UAC1 adaptive, not tested with UAC1 async.

Reported-by: B. Zhang <bb.zhang@free.fr>
Signed-off-by: Torstein Hegge <hegge@resisty.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-05-17 08:05:34 +02:00
Eldad Zack
4ca231b2e6 ALSA: usb-audio: caiaq: fix endianness bug in snd_usb_caiaq_maschine_dispatch
Current code does this:

  be16_to_cpu(buf[i * 2] << 8 | buf[(i * 2) + 1])

Which is effectively (neglecting the index):

  be16_to_cpu(be16_to_cpu(*((u16 *) buf)))

This means the int16 in the buffer is not converted at all.

Daniel Mack confirmed that the driver works on little endian
CPUs, leading to the conclusion that the device-side structure
is actually little endian.
This changes the code to use le16_to_cpu().

Caught by sparse.

Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-30 09:19:02 +02:00
Eldad Zack
74c34ca1cc ALSA: pcm_format_to_bits strong-typed conversion
Add a function to handle conversion from snd_pcm_format_t
to bitwise with proper typing.

Change such conversions to use this function and silence sparse
warnings.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-29 13:36:15 +02:00
Clemens Ladisch
c75c5ab575 ALSA: USB: adjust for changed 3.8 USB API
The recent changes in the USB API ("implement new semantics for
URB_ISO_ASAP") made the former meaning of the URB_ISO_ASAP flag the
default, and changed this flag to mean that URBs can be delayed.
This is not the behaviour wanted by any of the audio drivers because
it leads to discontinuous playback with very small period sizes.
Therefore, our URBs need to be submitted without this flag.

Reported-by: Joe Rayhawk <jrayhawk@fairlystable.org>
Cc: <stable@vger.kernel.org> # 3.8 only
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-29 10:57:35 +02:00
David Henningsson
fa92dd77ec ALSA: usb - Avoid unnecessary sample rate changes on USB 2.0 clock sources
The Scarlett 2i2 seems to take almost 500 ms to set the sample rate,
even if the clock is currently set to that value. This patch speeds
up prepare of the device, by avoiding setting the clock to something
it already is.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-26 07:37:09 +02:00
Trulan Martin
03e0221444 ALSA: usb-audio: USB quirk for Yamaha THR10C
This patch adds a USB quirk for the Yamaha THR10C amp.

Signed-off-by: Trulan Martin <trulanm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:48:21 +02:00
Trulan Martin
1b15362c74 ALSA: usb-audio: USB quirk for Yamaha THR5A
This patch adds a USB quirk for the Yamaha THR5A amp.

Signed-off-by: Trulan Martin <trulanm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:48:02 +02:00
Trulan Martin
ae3f0c267f ALSA: usb-audio: USB quirk for Yamaha THR10
This patch adds a USB quirk for the Yamaha THR10 amp.

Signed-off-by: Trulan Martin <trulanm@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:47:50 +02:00
Takashi Iwai
60af3d037e ALSA: usb-audio: Fix autopm error during probing
We've got strange errors in get_ctl_value() in mixer.c during
probing, e.g. on Hercules RMX2 DJ Controller:

  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x201, wIndex = 0xa00, type = 4
  ALSA mixer.c:352 cannot get ctl value: req = 0x83, wValue = 0x200, wIndex = 0xa00, type = 4
  ....

It turned out that the culprit is autopm: snd_usb_autoresume() returns
-ENODEV when called during card->probing = 1.

Since the call itself during card->probing = 1 is valid, let's fix the
return value of snd_usb_autoresume() as success.

Reported-and-tested-by: Daniel Schürmann <daschuer@mixxx.org>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:46:51 +02:00
Daniel Mack
ebfc594c02 ALSA: snd-usb: try harder to find USB_DT_CS_ENDPOINT
The USB_DT_CS_ENDPOINT class-specific endpoint descriptor is usually
stuffed directly after the standard USB endpoint descriptor, and this is
where the driver currently expects it to be.

There are, however, devices in the wild that have it the other way
around in their descriptor sets, so the USB_DT_CS_ENDPOINT comes
*before* the standard enpoint. Devices known to implement it that way
are "Sennheiser BTD-500" and Plantronics USB headsets.

When the driver can't find the USB_DT_CS_ENDPOINT, it won't be able to
change sample rates, as the bitmask for the validity of this command is
storen in bmAttributes of that descriptor.

Fix this by searching the entire interface instead of just the extra
bytes of the first endpoint, in case the latter fails.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Reported-and-tested-by: Torstein Hegge <hegge@resisty.net>
Reported-and-tested-by: Yves G <alsa-user@vivigatt.com>
Cc: stable@kernel.org
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-25 07:33:20 +02:00
Daniel Schürmann
b5f035dbca ALSA: snd-usb-audio: set the timeout for usb control set messages to 5000 ms
Set the timeout for USB control set messages according to the USB 2
spec, using the macros from include/linux/usb.h.
The get timout becomes 5000 ms even though it is 500 ms in the
spec. This patch is required to run the Hercules RMX2 which needs a
timeout of 1240 ms.

More notes from author:
I still distinguish between set and get but as long both are 5000 ms
GCC will remove it anyway. IMHO this is more easy read and there is no
need to explain why we use a get timeout for set messages.

Signed-off-by: Daniel Schürmann <daschuer@mixxx.org>
Acked-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-22 10:45:02 +02:00
Takashi Iwai
8dd2b66d1a ASoC: More updates for v3.10
The main additional change here is Lars-Peter's DMA work plus the
 platform conversions which have been tested - getting this in mainline
 will make life easier for development after the merge window.  These
 factor a large chunk of code out of the drivers for the platforms using
 dmaengine, greatly simplifying development.
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Merge tag 'asoc-v3.10-2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next

ASoC: More updates for v3.10

The main additional change here is Lars-Peter's DMA work plus the
platform conversions which have been tested - getting this in mainline
will make life easier for development after the merge window.  These
factor a large chunk of code out of the drivers for the platforms using
dmaengine, greatly simplifying development.
2013-04-18 16:24:31 +02:00
Daniel Mack
126825e7ea ALSA: snd-usb: add quirks handler for DSD streams
Unfortunately, none of the UAC standards provides a way to identify DSD
(Direct Stream Digital) formats. Hence, this patch adds a quirks
handler to identify USB interfaces that are capable of handling DSD.

That quirks handler can augment the already parsed formats bit-field,
by any of the new SNDRV_PCM_FMTBIT_DSD_{U8_U16} and setting the dsd_dop
flag in the audio format, if the driver should take care for the DOP
byte stuffing.

The only devices that are known to work with this are the ones with
a 'Playback Designs' vendor id.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:53 +02:00
Daniel Mack
44dcbbb1cd ALSA: snd-usb: add support for bit-reversed byte formats
There is quite some confusion around the bit-ordering in DSD samples,
and no general agreement that defines whether hardware is supposed to
expect the oldest sample in the MSB or the LSB of a byte.

ALSA will hence set the rule that on the software API layer, bytes
always carry the oldest bit in the most significant bit of a byte, and
the driver has to translate that at runtime in order to match the
hardware layout.

This patch adds support for this by adding a boolean flag to the
audio format struct.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:47 +02:00
Daniel Mack
d24f5061ee ALSA: snd-usb: add support for DSD DOP stream transport
In order to provide a compatibility way for pushing DSD
samples through ordinary PCM channels, the "DoP open Standard" was
invented. See http://www.dsd-guide.com for the official document.

The host is required to stuff DSD marker bytes (0x05, 0xfa,
alternating) in the MSB of 24 bit wide samples on the bus, in addition
to the 16 bits of actual DSD sample payload.

To support this, the hardware and software stride logic in the driver
has to be tweaked a bit, as we make the userspace believe we're
operating on 16 bit samples, while we in fact push one more byte per
channel down to the hardware.

The DOP runtime information is stored in struct snd_usb_substream, so
we can keep track of our state across multiple calls to
prepare_playback_urb_dsd_dop().

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:32 +02:00
Daniel Mack
8a2a74d2b7 ALSA: snd-usb: use ep->stride from urb callbacks
For normal PCM transfer, this change has no effect, as the endpoint's
stride is always frame_bits/8. For DSD DOP streams, however, which is
added later, the hardware stride differs from the software stride, and
the endpoint has the correct information in these cases.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-18 10:03:23 +02:00
Clemens Ladisch
cbc200bca4 ALSA: usb-audio: disable autopm for MIDI devices
Commit 88a8516a21 (ALSA: usbaudio: implement USB autosuspend)
introduced autopm for all USB audio/MIDI devices.  However, many MIDI
devices, such as synthesizers, do not merely transmit MIDI messages but
use their MIDI inputs to control other functions.  With autopm, these
devices would get powered down as soon as the last MIDI port device is
closed on the host.

Even some plain MIDI interfaces could get broken: they automatically
send Active Sensing messages while powered up, but as soon as these
messages cease, the receiving device would interpret this as an
accidental disconnection.

Commit f5f165418c (ALSA: usb-audio: Fix missing autopm for MIDI input)
introduced another regression: some devices (e.g. the Roland GAIA SH-01)
are self-powered but do a reset whenever the USB interface's power state
changes.

To work around all this, just disable autopm for all USB MIDI devices.

Reported-by: Laurens Holst
Cc: <stable@vger.kernel.org>
Signed-off-by: Clemens Ladisch <clemens@ladisch.de>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-15 16:03:57 +02:00
Calvin Owens
1539d4f82a ALSA: usb: Add quirk for 192KHz recording on E-Mu devices
When recording at 176.2KHz or 192Khz, the device adds a 32-bit length
header to the capture packets, which obviously needs to be ignored for
recording to work properly.

Userspace expected:  L0 L1 L2 R0 R1 R2
...but actually got: R2 L0 L1 L2 R0 R1

Also, the last byte of the length header being interpreted as L0 of
the first sample caused spikes every 0.5ms, resulting in a loud 16KHz
tone (about the highest 'B' on a piano) being present throughout
captures.

Tested at all sample rates on an E-Mu 0404USB, and tested for
regressions on a generic USB headset.

Signed-off-by: Calvin Owens <jcalvinowens@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-13 10:58:03 +02:00
Daniel Mack
21bb5aafce ALSA: snd-usb: Playback Design: use usb_set_inferface quirk from more locations
It turns out the devices from Playback Design need the delay quirk
after usb_set_interface from clocks.c as well. Make it a proper
quirks function and factor out the code to quirks.c.

Signed-off-by: Daniel Mack <zonque@gmail.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-10 09:21:43 +02:00
Eldad Zack
889d66848b ALSA: usb-audio: fix endianness bug in snd_nativeinstruments_*
The usb_control_msg() function expects __u16 types and performs
the endianness conversions by itself.
However, in three places, a conversion is performed before it is
handed over to usb_control_msg(), which leads to a double conversion
(= no conversion):
* snd_usb_nativeinstruments_boot_quirk()
* snd_nativeinstruments_control_get()
* snd_nativeinstruments_control_put()

Caught by sparse:

sound/usb/mixer_quirks.c:512:38: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:512:38:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:512:38:    got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:35: warning: incorrect type in argument 5 (different base types)
sound/usb/mixer_quirks.c:543:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/mixer_quirks.c:543:35:    got restricted __le16 [usertype] <noident>
sound/usb/mixer_quirks.c:543:56: warning: incorrect type in argument 6 (different base types)
sound/usb/mixer_quirks.c:543:56:    expected unsigned short [unsigned] [usertype] index
sound/usb/mixer_quirks.c:543:56:    got restricted __le16 [usertype] <noident>
sound/usb/quirks.c:502:35: warning: incorrect type in argument 5 (different base types)
sound/usb/quirks.c:502:35:    expected unsigned short [unsigned] [usertype] value
sound/usb/quirks.c:502:35:    got restricted __le16 [usertype] <noident>

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Acked-by: Daniel Mack <zonque@gmail.com>
Cc: <stable@vger.kernel.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-07 09:44:08 +02:00
Eldad Zack
1dc669fed6 ALSA: usb-audio: UAC2: support read-only freq control
Some clocks might be read-only, e.g., external clocks (see also
UAC2 4.7.2.1).

In this case, setting the sample frequency will always fail
(even if the rate is equal to the current clock rate),
therefore do not write, but read the value and compare to the
requested rate.
If the clock is read only, avoid reading it twice.

If it doesn't match, return -ENXIO since the clock is invalid for
this configuration.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:32:07 +02:00
Eldad Zack
027bbc1546 ALSA: usb-audio: show err in set_sample_rate_v2 debug
Show the error code returned from the USB subsystem in
the debug messages.

Signed-off-by: Eldad Zack <eldad@fogrefinery.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
2013-04-04 08:31:40 +02:00